On one of our locations, I am having issues with one-way audio when I call
several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the
phones individually, they work fine, so it's not a volume setting on the
phone. Also this setup has worked at other locations. Any idea's what to look for? Thanks in advance.
On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote:
> There were a few compatability issues with the tdm400's pci interface chip
> and certain motherboards Interesting idea, but it's been running in this same server with the
same motherboard for at least two years now, and this problem started
only last week. At this point, I am sure enough that the card has failed that I have
ordered a new base card. If I'm wrong, I will only be out return postage
and 10% restock fee.
On 04/15/2012 07:26 PM, Patrick Lists wrote:
> On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
>> Is it a good idea to use asterisk transcoding from G711 to iLBC or
>> should I find out any other solution not involving transcoding (f.e.
>> using G.729 that is supported in both sides). I'm worried about voice
>> quality and trying to avoid paying for G.729 licensing.
>> Anybody with experience or quantitative measurements of the voice
>> quality degradation in that scenario?
> The term that may interest…
I am struggling to get the mac-addresses of IP phones that are connected
to asterisk as the phone are in different VLAN with * and they were
manually configured. I want to centralize their configuration using
res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? Sam
First off all let me apologize for my English.
My company use an Avaya PBX with one E1 and about 250 extensions
(phones, faxes alarms, visa card readers etc).
My idea was to install a transparent asterisk between TELCO and PBX. I use an OpenVox DE210E.
After install Elastix 2.2.0 and run dahdi_genconfig, i edit
/etc/asterisk/dahdi-channels.conf changing span 1 group from 0,11 to 0
and span 2 from 0,12 to 1. Then from gui add all the active avaya extensions as custom extension
with dial DAHDI/i2/extension-number The phones works fine but faxes…
I'm looking at replacing a PBX for a small business with an asterisk
box. I'm rather attracted to the idea of one of the iso distributions
where someone did most of the integration for us already ;) Can anyone comment on the pros/cons of the various options? I'm seeing
several options out there: -Trixbox CE (no new version since 2010? is this project dead?)
-PBX in a Flash
we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of:
ast_log(LOG_WARNING, "Unable to leave message since we will exceed the maximum number of messages allowed (%u >= %u)n", msgnum, vmu->maxmsg);
but I was wondering how feasible it would be to modify the code to add:
1) the mailbox name of the user whom has hit the limit
2) a warning/critical threshold that the user is…
I have an asterisk box which has Polycom Soundpoints IP335 and IP650s registering to it both locally and remote.
I want to be able to incorporate a cordless phone at the remote location; not a wireless phone.
I want it to also be able to register to the same asterisk box so it can take calls and transfers.
I have been running the windows vovida stun server for some time and has
worked without issue, but I really want to run a linux stun server and get
away from the windows based one. Anyone have an idea of a good replacment
that can be compled on opensuse?